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Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK 1.2



 

Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK 1.2

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Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK 1.2 Ranking & Summary

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User Review: 0 (0 times)
File size: 7.02 MB
Platform: WinXP,WinVista,WinVista x64,Win7 x32,Win7 x64
License: Commercial
Price: $999
Downloads: 24
Date added: 2010-02-10

Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK 1.2 description


Softphone SDK, SIP SDK, VOIP SDK, Softphone, IVR SDK, IMS SDK.
Our brand-new SIP Client SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications and websites. It accelerates the development of SIP/RTP compliant soft phone with a fully-customizable user interface and brand name.
The conaito VoIP SIP Client SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGC), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!
Key features
- Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider
- VoIP conferencing with crystal clear sound even for both low and high-bandwidth users
(G711 A-Law, G711 U-Law, Speex, GSM6.10, iLBC, L16 and g729 Codec)
- Multi-party voice conference support
- Multi-line support (Multiple concurrent calls)
- Line Hold/Retrieve support
- Call Transfer support
- Mute microphone/speaker
- Auto-answer
- Do Not Disturb (DND)
- Adaptive Jitter buffer
- PLC (Packet Lost Concealment)
- AGC (auto gain controller)
- AES (Acoustic echo cancellation or suppression)
- Noise cancellation or suppression
- DTMF tones support (generation/detection)
- Recording and play voice conversation into PCM WAVE (.wav) file
...and much more. Try it today!

System Requirements: For .NET and all ActiveX

Enhancements: *Easily make and receive SIP based phone calls * Multi-Line, Multi-User conference support * Make and receive SIP based phone calls * VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729) * STUN, DTMF and Call Transfer

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